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Author SHA1 Message Date
Guillaume Desmottes
75dc98cc08 h265parse: set interlace-mode=interleaved on interlaced content 
interlace-mode=alternate is a special case of interlace-mode=interleaved
where the fields are split using two different buffers.

We should use the latter instead of the former to no break compat with
elements supporting only 'interleaved'.
Decoders producing alternate, such as OMX on the Zynq, should change the
interlace-mode on their output caps.

Fix https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/825

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1655>
 2020-10-09 10:19:52 +00:00
Jacek Tomaszewski
ca4a0273df Replace LGPL v2 with LGPL v2.1 in COPYING and remove COPYING.LIB 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1631>
 2020-10-09 08:48:17 +00:00
Jacek Tomaszewski
9cac8bb449 Replace GPL v2 with LGPL v2 in COPYING file 
Fixes #1422
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1422

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1631>
 2020-10-09 08:48:17 +00:00
Jan Alexander Steffens (heftig)
c6eeead1e4 srt: Consume the error from gst_srt_object_write 
Instead of leaking it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1668>
 2020-10-09 07:47:47 +00:00
Jan Alexander Steffens (heftig)
2a7fa67693 srt: Check socket state before retrieving payload size 
The connection might be broken, which we should detect instead of just
aborting the write.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1669>
 2020-10-09 07:12:04 +00:00
Jakub Adam
6f2f15b5fb x265enc: fix deadlock on reconfig 
Don't attempt to obtain encoder lock that is already held by
gst_x265_enc_encode_frame().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1671>
 2020-10-09 06:39:36 +00:00
Sebastian Dröge
d3d73f61fa webrtc: Require gstreamer-sdp in the pkg-config file 
Some headers include API from it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1660>
 2020-10-08 21:55:18 +00:00
Edward Hervey
dd11e91c3b srtsrc: Fix timestamping 
SRT provides the original timestamp of a packet (with drift/skew corrected for
local clock), which is what should be used for timestamping the outgoing
buffers. This ensures that we output the packets with the same timestamp (and by
extension rate) as the original feed.

Also detect if packets were dropped (by checking the sequence number) and
properly set DISCONT flag on the outgoing buffer.

Finally answer the latency queries

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1658>
 2020-10-08 21:12:17 +00:00
Seungha Yang
20d9283e3d mfvideosrc: Use only the first video stream per device 
Non-first video stream might not be working with current
implementation. It could be non-video (e.g., photo source) and then
ReadSample() might be blocked forever.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1661>
 2020-10-08 20:43:58 +00:00
Seungha Yang
9279326d8a decklink: Update doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
b86e77e3a3 decklink: Update Windows headers with SDK 11.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
94a9a8f836 decklink: Update OSX headers with SDK 11.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Tim
c6151f635f decklink: Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro 
Updated Decklink SDK to version 11.2 in order to support newer cards like the Decklink 8K Pro.
This required to replace the duplex property by a profile property.

Profile values can be the following:
-  bmdProfileOneSubDeviceFullDuplex
-  bmdProfileOneSubDeviceHalfDuplex
-  bmdProfileTwoSubDevicesFullDuplex
-  bmdProfileTwoSubDevicesHalfDuplex
-  bmdProfileFourSubDevicesHalfDuplex

Fixes #987

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
 2020-10-08 20:05:03 +00:00
Seungha Yang
9ecdfea7da mfvideosrc: Fix invalid memory access when outputting jpeg 
Don't access unknown-dangerous-nonsense address

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1662>
 2020-10-08 16:28:16 +00:00
Sebastian Dröge
cc7e98816f Revert "webrtc: Save the media kind in the transceiver" 
This reverts commit f54d8e9945.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:12 +03:00
Sebastian Dröge
a40d6f4994 Revert "rtpsender: Add API to set the priority" 
This reverts commit a8b287c764.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:10 +03:00
Sebastian Dröge
849839ba97 Revert "rtptransceiver: Store the SSRC of the current stream" 
This reverts commit d1da271f25.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:07 +03:00
Sebastian Dröge
e65a8cbcf1 Revert "webrtcbin: Remove unused function" 
This reverts commit 39723dbe93.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:04 +03:00
Sebastian Dröge
b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority" 
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:00 +03:00
Sebastian Dröge
f12265d9c5 Revert "webrtc: Document more objects" 
This reverts commit ad68c6b1eb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:52:50 +03:00
Sebastian Dröge
74a42c5ba8 Revert "webrtc: Add hotdoc style since tags" 
This reverts commit 63a5fa818c.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:51:37 +03:00
Olivier Crête
63a5fa818c webrtc: Add hotdoc style since tags 
We're stuck having to add a separate comment for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:52:48 -04:00
Olivier Crête
ad68c6b1eb webrtc: Document more objects 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
8ba08598bb webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
39723dbe93 webrtcbin: Remove unused function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
d1da271f25 rtptransceiver: Store the SSRC of the current stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
a8b287c764 rtpsender: Add API to set the priority 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Olivier Crête
f54d8e9945 webrtc: Save the media kind in the transceiver 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
Jan Alexander Steffens (heftig)
92dc2f4192 srt: Remove unused sa_family tracking 
Now that SRT no longer needs the family when creating the socket, this
code has become useless.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Niklas Hambüchen
13c8bda531 srt: Move off deprecated srt_socket(). 
See 73ee1e1a3e/docs/API-functions.md (srt_socket)

`srt_create_socket()` was added in
4b897ba92d
and srt `v1.3.0` is the first release that has it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 13:56:32 +02:00
Jan Alexander Steffens (heftig)
4e26b447f6 srt: Register a log handler 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:04 +02:00
Jan Alexander Steffens (heftig)
936f422764 srt: Avoid removing invalid sockets from the polls 
This would provoke error messages from SRT.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:39:00 +02:00
Jan Alexander Steffens (heftig)
fda4cfd15e srt: Fix use of srt_startup 
`srt_startup` can also return 1 if it was successful. Avoid warning in
this case.

Avoid a race when checking whether we need to call it at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:38:57 +02:00
Jan Alexander Steffens (heftig)
6b8c4a5f34 srt: Fix parameter types used for socket options 
The [SRT documentation][1] specifies exact types for the socket options.
Make sure we match these.

This reverts the linger workaround in commit 84f8dbd932
and extends srt_constant_params to support other types than int.

[1]: https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1659>
 2020-10-06 12:36:40 +02:00
Seungha Yang
0a454191c0 d3d11upload: Allow passthrough for system memory 
... like how d3d11download and gl{upload,download} do.
This should've been part of the commit 9b72b04dad
but I missed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1645>
 2020-10-05 15:54:37 +00:00
Víctor Manuel Jáquez Leal
dcc4557dd6 va: allocator: refactor flush methods for both allocators 
Since the logic is the same, it can be generalized in a single common
function.

Also the methods run the common function with a lock and signal the
buffers' conditional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
37fa6df57d va: allocator: refactor GstVaDmabufAllocator 
Move code down to group it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
9c56c1b3e8 va: allocator: refactor GstVaBuffersurface 
Move code up and add namespace to methods, and renaming
_creating_buffer_surface() to the canonical
gst_va_buffer_surface_new()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
734e2a74c4 va: implement pooled allocators 
1. Allocators don't implement memory free() methods since all the memories will
   implement dispose() returning FALSE
2. Memory/miniobject dispose() will act as memory release, enqueueing the
   release memory
3. A new allocator's method prepare_buffer() which queries the released memory
   queue and will add the requiered memories to the buffer.
4. Allocators added a GCond to synchronize dispose() and prepare_buffer()
5. A new allocator's method flush() which will free for real the memories.

While the bufferpool will

1. Remove all the memories at reset_buffer()
2. Implement acquire_buffer() calling allocator's prepare_buffer()
3. Implement flush_start() calling allocator's flush()
4. start() is disabled since it pre-allocs buffers but also calls
   our reset_buffer() which will drop the memories and later the
   buffers are ditched, something we don't want. This approach avoids
   buffer pre-allocation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
d6f9cfc159 va: allocator: user gst_clear_object() for _buffer_surface_unref() 
Event if this function is only used by gst_va_dmabuf_memories_setup(), it might
get reused later by GstVaDmabufAllocator's functions. This change makes the
function less fragile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
10b3250175 va: allocator: renamed gst_va_dmabuf_memory_release() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
1617dcbe9e va: allocator: renamed available_mems queue 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
303b1ea350 va: allocator: rename gst_va_dmabuf_allocator_setup_buffer() 
Since it's related with GstVaDmabufAllocator.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
bb586b227f va: allocator: calculated surface frame internally 
Instead of using gst_buffer_get_size() just add the memory sizes reported by
exported fd.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
864f0c2ede va: allocator: make GstVaMemory shareable 
Renamed the first variable member of GstVaMemory from parent to mem in
order to avoid confusion with GstMemory's parent.

When freeing the structure, memory's parent is check in order to
decide if surfaces has to be destroyed or not, since only the parent
class have to destroy it.

Removed GST_MEMORY_FLAG_NO_SHARE in memory initialization, since it is
deprecated.

Implemented allocator's share virtual method which creates a new
shallow GstVaMemory structure based on the passed one which will be
it's parent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
0b6d8e73ec va: allocator: remove copy method for GstVaMemory 
Since the memory has to be shareable. That will be address in the next
commits.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Víctor Manuel Jáquez Leal
43b950b252 va: allocator: remove va allocator mem_is_span() vmethod 
Since it is the default by base class.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1626>
 2020-10-04 18:42:51 +00:00
Seungha Yang
a9314b4554 d3d11upload/d3d11download: Make use of staging buffer 
... instead of direct cpu map for d3d11memory object. In this way,
we don't need per GstD3D11Memory staging texture.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1627>
 2020-10-04 15:44:03 +00:00
Seungha Yang
f35eeaa02b d3d11: Don't hold staging texture 
Staging texture is used for memory transfer between system and
gpu memory. Apart from d3d11{upload,download} elements, however,
it should happen very rarely.

Before this commit, d3d11bufferpool was allocating at least one
staging texture in order to calculate cpu accessible memory size,
and it wasn't freed for later use of the texture unconditionally.
But it will increase system memory usage. Although GstD3D11memory
object is implemented so that support CPU access, most memory
transfer will happen in d3d11{upload,download} elements.

By this commit, the initial staging texture will be freed immediately
once cpu accessible memory size is calculated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1627>
 2020-10-04 15:44:03 +00:00
Seungha Yang
899f9978f5 d3d11: Move to G_DECLARE_FINAL_TYPE 
... and remove unnecessary forward declaration.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1627>
 2020-10-04 15:44:03 +00:00